AES San Francisco 2010Paper Session P17

Saturday, November 6, 2:30 pm — 6:30 pm (Room 220)

Paper Session: P17 - Real-Time Audio Processing

Chair:Jayant Datta

P17-1 A Time Distributed FFT for Efficient Low Latency Convolution—Jeffrey Hurchalla, Garritan Corp. - Orcas, WA, USA
To enable efficient low latency convolution, a Fast Fourier Transform (FFT) is presented that balances processor and memory load across incoming blocks of input. The proposed FFT transforms a large block of input data in steps spread across the arrival of smaller blocks of input and can be used to transform large partitions of an impulse response and input data for efficiency, while facilitating convolution at very low latency. Its primary advantage over a standard FFT as used for a non-uniform partition convolution method is that it can be performed in the same processing thread as the rest of the convolution, thereby avoiding problems associated with the combination of multithreading and near real-time calculations on general purpose computing architectures.
Convention Paper 8257 (Purchase now)

P17-2 An Infinite Impulse Response (IIR) Hilbert Transformer Filter Design Guide for Audio—Daniel Harris, Sennheiser Research Laboratory - Palo Alto, CA, USA; Edgar Berdahl, Stanford University - Stanford, CA, USA
Hilbert Transformers have found many applications in the signal processing community, from single-sideband communication systems to audio effects. IIR implementations are attractive for computationally sensitive systems due to their lower number of coefficients. However, as in any advanced filter design problem, their tuning and implementation present a number of design challenges and tradeoffs. Furthermore, while literature addressing these problems exists, designers must draw from several sources to find answers. In this paper we present a complete start-to-finish explanation of how to implement an efficient infinite impulse response (IIR) Hilbert transformer filter. We start from a half-band filter design and show how the poles move as the half-band filter is transformed into summed all-pass filters and then from there into a Hilbert transformer filter. The design technique is based largely on pole locations and creates a filter in the cascaded 1st order allpass form, which is numerically robust.
Convention Paper 8258 (Purchase now)

P17-3 Automatic Parallelism from Dataflow Graphs—Ramy Sadek, University of Southern California - Playa Vista, CA, USA
This paper presents a novel algorithm to automate high-level parallelization from graph-based data structures representing data flow. Algorithm correctness is shown via a formal proof by construction. This automatic optimization yields large performance improvements for multi-core machines running host-based
applications. Results of these advances are shown through their incorporation into the audio processing engine Application Rendering Immersive Audio (ARIA) presented at AES 117. Although the ARIA system is the target framework, the contributions presented in this paper are generic and therefore applicable in a variety of software such as Pure Data and Max/MSP, game audio engines, non-linear editors and related systems. Additionally, the parallel execution paths
extracted are shown to give effectively optimal cache performance, yielding significant speedup for such host-based applications.
Convention Paper 8259 (Purchase now)

P17-4 The Design of Low-Complexity Wavelet-Based Audio Filter Banks Suitable for Embedded Platforms—Neil Smyth, CSR - Cambridge Silicon Radio - Belfast, N. Ireland, UK
Many audio applications require the use of low complexity, low power, and low latency filter banks (e.g., real-time audio streaming to mobile devices). The underlying mathematics of wavelet transforms provides these attractive characteristics for embedded platforms. However, commonly used wavelets (Haar, Daubechies) possess coefficients containing irrational numbers that lead to distortion in fixed-point implementations. This paper discusses the development and provides practical performance comparisons of filter banks using wavelet transforms as an alternative to more commonly used sub-banding filter banks in PCM audio coding algorithms. The advantages and disadvantages of wavelets used in such audio compression applications are also discussed.
Convention Paper 8260 (Purchase now)

P17-5 Application of Optimized Inverse Filtering to Improve Time Response and Phase Linearization in Multiway Loudspeaker Systems—Mario Di Cola, Audio Labs Systems - Casoli (CH), Italy; Daniele Ponteggia, Studio Ing. Ponteggia - Terni (TR), Italy
Digital processing has been widely demonstrated to be a very useful technique in improving loudspeaker systems’ performances. Particularly interesting is Inverse Filtering applied to loudspeaker systems because it can improve performances and sound quality in terms of transient response and reduced overall phase shift. Inverse Filtering is a processing technique that can be realized with FIR filtering techniques with a specific sequence of taps that need to be synthesized “ad hoc” for a specific transducer and/or for a specific loudspeaker system configuration. Most of the studies on this matter so far, with very few exceptions, have been focused on the “DSP processing” point of view, being generally related to the involved mathematics and relative numerical problems. This paper represents a discussion on the philosophy that should drive the application of this technique to process a loudspeaker system in order to really improve it, and consequently it’s been focused on the analysis of the loudspeaker system nature and the understanding of what can really be processed with a 1-dimensional “action.” We will discuss what can be synthesized as a “2-port” model of the loudspeaker and then what can be effectively obtained by processing the input signal of a loudspeaker system.
Convention Paper 8261 (Purchase now)

P17-6 Filter Design for a Double Dipole Flat Panel Loudspeaker System Using Time Domain Toeplitz Equations—Tobias Corbach, Martin Holters, Udo Zölzer, Helmut-Schmidt-University/University of the Federal Armed Forces - Hamburg, Germany
Today flat panel loudspeakers are used in multiple applications. Due to their high directivity and their good structural integration properties, flat panel loudspeakers are commonly used for directed acoustic information. A previously proposed system of 2 parallel flat panel dipole loudspeakers with adapted input filtering ensures a high suppression of the backward radiation and only minor influences to the forward radiation side. This paper presents a new approach to the filter computation for this application. It makes use of the time domain convolution, realized by Toeplitz matrices and builds the desired filter impulse responses by a least squares approach. The different filter computations as well as the numerical and measured results are shown.
Convention Paper 8262 (Purchase now)

P17-7 A Low Complexity Approach for Loudness Compensation—Pradeep D. Prasad, Ittiam Systems Pvt. Ltd. - Bangalore, Karnataka, India
The essence of loudness compensation is to maintain the perceived spectral balance of audio content irrespective of the playback volume level. The need for this compensation arises due to the inherent non-linearity in human aural perception manifesting as change in spectral balance. The compensation varies with critical band, original, and playback specific loudness. This results in a computationally intensive approach of estimating original and target specific loudness and calculating required compensation for every frame. A low complexity algorithm is proposed to enable resource constrained devices to efficiently perform loudness compensation. A closed form expression is derived for the proposed compensation followed by an analysis of the quality versus complexity tradeoff.
Convention Paper 8263 (Purchase now)